diff options
| author | kpfleming <kpfleming@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-08-03 20:58:48 +0000 |
|---|---|---|
| committer | kpfleming <kpfleming@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-08-03 20:58:48 +0000 |
| commit | 538f4ca207028ee047ffadc741278584c2ca43bc (patch) | |
| tree | 80d368e550777a4a58bfd45fded1fad152c9b5fc /main/rtp.c | |
| parent | b78e0c85c91f64adf0a9f023e302e9e38038d7b7 (diff) | |
Merged revisions 210190 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
........
r210190 | kpfleming | 2009-08-03 15:48:48 -0500 (Mon, 03 Aug 2009) | 11 lines
Rename 'canreinvite' option to 'directmedia', with backwards compatibility.
It is clear from multiple mailing list, forum, wiki and other sorts of posts
that users don't really understand the effects that the 'canreinvite' config
option actually has, and that in some cases they think that setting it to 'no'
will actually cause various other features (T.38, MOH, etc.) to not work properly,
when in fact this is not the case. This patch changes the proper name of the
option to what it should have been from the beginning ('directmedia'), but
preserves backwards compatibility for existing configurations.
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@210191 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'main/rtp.c')
| -rw-r--r-- | main/rtp.c | 4 |
1 files changed, 2 insertions, 2 deletions
diff --git a/main/rtp.c b/main/rtp.c index c77db1969..1ae21bab5 100644 --- a/main/rtp.c +++ b/main/rtp.c @@ -2124,7 +2124,7 @@ int ast_rtp_early_bridge(struct ast_channel *c0, struct ast_channel *c1) text_src_res = srcpr->get_trtp_info ? srcpr->get_trtp_info(c1, &tsrcp) : AST_RTP_GET_FAILED; } - /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */ + /* Check if bridge is still possible (In SIP directmedia=no stops this, like NAT) */ if (audio_dest_res != AST_RTP_TRY_NATIVE || (video_dest_res != AST_RTP_GET_FAILED && video_dest_res != AST_RTP_TRY_NATIVE)) { /* Somebody doesn't want to play... */ ast_channel_unlock(c0); @@ -2211,7 +2211,7 @@ int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, i else destcodec = 0; - /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */ + /* Check if bridge is still possible (In SIP directmedia=no stops this, like NAT) */ if (audio_dest_res != AST_RTP_TRY_NATIVE || (video_dest_res != AST_RTP_GET_FAILED && video_dest_res != AST_RTP_TRY_NATIVE) || audio_src_res != AST_RTP_TRY_NATIVE || (video_src_res != AST_RTP_GET_FAILED && video_src_res != AST_RTP_TRY_NATIVE) || !(srccodec & destcodec)) { /* Somebody doesn't want to play... */ ast_channel_unlock(dest); |